RtpSender uses a transactional model when getting and setting RtpParameters.
One must call GetParameters() and then can use the returned object in a
subsequent call to SetParameters().
PeerConnection was calling GetParameters() and SetParameters() during
negotiation in SetLocalDescription and SetRemoteDescription effectively
invalidating any parameters that the client previously held.
This change introduces an internal way for the platform to modify
parameters without invalidating the transactional model, provided that
the modification is not severe.
Ex. removing simulcast layers is a severe modification and will
invalidate any outstanding parameters.
Bug: webrtc:10339
Change-Id: I362e8ca4d9556e04a1aa7a3e74e2c275f8d16fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/124504
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26864}
Most of the implementation in rtp_sender.cc is a copy paste for both
Audio & Video RTP senders. This change moves all the common behavior
into the base RtpSenderInternal class.
Template method pattern is used to accomodate for the very slight differences
between audio and video senders.
Bug: None
Change-Id: I6d4e93cd32fbb0fb361fd0e1883791019bde9a92
Reviewed-on: https://webrtc-review.googlesource.com/c/123411
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26758}
Added a layer in RtpSender that bridges the gap between the layers
that the user sees and the layer that the media engine sees.
Media engine still maintains the invariant that the number of layers
cannot be changed, while RtpSender adds and removes layers between
the user GetParameters and SetParameters calls and the media engine.
Bug: webrtc:10251
Change-Id: I33839c1f9a9052cb6130253e5a582606f2cbe54a
Reviewed-on: https://webrtc-review.googlesource.com/c/122641
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26756}
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
